Medium and even small size enterprises which rely on inbound voice calls have in the past relied on landline techniques and PBX exchanges at the premises in order to handle the incoming calls. Not only are PBX exchanges costly, the PBX technology is presently outdated and does not support many of the functions that are required for the taking of the inbound calls. Not only are the landlines expensive, the landline tariff structure for calls to various area codes oftentimes requires long-distance packages which are cost prohibitive. Establishments such as pizza delivery services, restaurants, automobile dealerships, auto service centers, real estate offices and the like require personnel to take incoming calls on multiple lines into the establishment. It was thought that many of the cost problems associated with landlines could be solved using Voice over Internet Protocol, VoIP, systems.
The problem with many Voice over IP networks is the maintenance of quality voice communications when trying to route voice calls through an Internet protocol network due to the constant change in the quality of the connections as a result of network condition changes. Changes in communication channel conditions can result in packet loss, delay and jitter which ultimately affect the audio quality of the voice communication. In VoIP systems audio is converted from an analog form to digital packets which are transmitted over the VoIP network at which point they are transformed back into analog audio.
While VoIP systems regularly route calls through the networks based on available bandwidth, network overload is a problem in which packets are dropped after first having been delayed. The net result for the end-user is that voice quality degrades. The voice degradation ranges from undetectable to non-understandable.
These problems occur most severely in over-the-top service where there is no direct link to the public switched telephone network, PSTN. Because of the lack of dedicated links such as provided by major carriers including Comcast and other Multiple System Operators, MSOs, that have a dedicated quality of service in which customers are directly linked to the public switched telephone network, when Internet service providers are used as an intermediary to connect between the individual placing the call and the ultimate public switched telephone network, there have been complaints about the quality of connections that the Internet service provider provides. The Internet service providers referred to herein are cloud-based providers and the audio quality complained of is directly traceable to connections to the Internet service providers and how they, as an intermediary, connect the caller to the telephone network. These ISPs have not heretofore addressed packet loss, delay and jitter. Rather these ISPs are concerned with bandwidth and rely on codec compression techniques to achieve the lowest amount of bandwidth that one can possibly use for voice call. However, achieving the lowest bandwidth involves compression which itself degrades voice quality.
To the extent that VoIP providers seek to address voice quality at all, prior VoIP providers contract with the high quality data centers. By so doing they hope that the call path stays at a high quality. However, utilizing high-quality data centers is insufficient to address the aforementioned audio fidelity and connection problems.
There is therefore need to be able to route the calls to those servers having the least packet loss, delay or jitter and to do so in a robust fashion when using multiple servers in multiple data centers, so as to route the calls over the most robust channel.